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Troubleshooting
Below are some common issues and typical steps on troubleshooting/resolving them.
Before you begin those steps, here are some steps that should be performed beforehand:
Router Firmware Upgrade - Make sure you router has up-to-date firmware. Here are links to firmware downloads for some router manufacturers: Linksys
| D-Link | Netgear .Ping Test - This determines the packet loss and latency time to and from your destination and the quality of your network connection to our gateways.
You run the ping test by clicking on the "Start" button on your computer, click on "Run", type cmd and press "OK." This will open a command prompt window. Run the following command: ping sipdr.quantumvoice-sip.com
MOS Test - This resource helps you determine the suitability of your connection for VoIP by placing a call from wherever you are to selected destinations around the world and returning a MOS score.
For the MOS Test go here: MOS Test.
Speed Test - This test determines the speed of your internet connection: Speed Test
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If the below steps does not resolve the issue you are having, please contact us either via email customercare@quantumvoice.com at or by phone at 800-914-2943. Please also have the results of the above tests.
Service Issues
There is no dial tone. Fast Busy on outbound calls. I am experiencing dropped calls. My high speed Internet connection uses a USB port, not an Ethernet port. My phone does not ring / No incoming calls. Callers do not hear me while I am talking (one-way audio). I have echo on my line. I am hearing static on my line. Conversations are choppy sounding. What are latency, jitter, and packet loss and how do they affect my calls?Latency
Latency is the delay from when a packet leaves point A and when a response is returned from point B. For VOIP, this is the one-way delay between when something was spoken and when it was heard. The largest contributor to latency is caused by network transmission delay and round-trip latency affects dynamics of conversation.
With round trip latencies above 300 msec or so, users may experience annoying talk-over effects.
Jitter
Jitter refers to how variable latency is in a network. High jitter, greater than approximately 50 msec, can result in both increased latency and packet loss. Jitter causes packets to arrive at their destination with different timing and possibly in a different order than they were sent (spoken), with some arriving faster and some slower than they should.
To correct the effects of jitter, VOIP endpoints collect packets in a buffer and put them back together in the proper timing and order before the receiver hears them. Processing that buffer adds delay to the call, so the bigger the buffer, the longer the delay. If voice packets arrive when the buffer is full then packets are dropped and the receiver will never hear them. These are called discarded packets.
Packet Loss
Packet loss is when some of the voice packets are dropped by network routers or switches that become congested (lost packets), or discarded by the jitter buffer (discarded packets). This will affect the conversation in that you will miss words or parts of words in a conversation. This can often be caused by bad cabling or connection at the local-end or far-end, or problems on the internet backbone due to hardware failure or congestion.
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I get an "Out of Area" error message when dialing some toll-free numbers. I cannot send faxes through my ATA.Phone Adapter (ATA) Troubleshooting
What do the Status/Link LEDs indicate? How do I use the IVR menu (checking info through phone handset)?Phone Portal Troubleshooting
I can login to the portal but if I do anything I am asked to login again.
